Saturday, November 30, 2013

Tips & Techniques

If you know what you’re doing, plug‑ins are best
set by ear — but if you lack experience and take care to avoid rookie
mistakes, dialling in an off‑the‑peg preset can still prove effective...

Paul White

To really understand
the finer points of a Digital Audio Workstation, you need to know how
everything works in a traditional recording studio, including processors
such as reverb, echo, compression and EQ that now exist as plug‑ins
inside DAWs. Knowledge comes with time and experience, though, so ‘first
contact’ with a DAW can be intimidating. To make life easier,
manufacturers offer plug‑in presets that you can use in typical mixing
situations. Preset sounds for plug‑in instruments can often be used with
very little tweaking, but preset processor and effects settings are
more problematic, as how well they work for you will depend on things
like the level, frequency content and dynamic range of the signal, or
the project tempo. So how can you ensure they do the job with the
mimimum of intervention?

Crossing The Threshold

Preset names can look very appealing — but how could the patch designers know what material you planned to put through them?

During our Studio SOS visits and Mix Rescue sessions,
we often see DAW projects using a compressor preset that’s not
compressing at all! The reason is obvious if you know about compressors,
but for the rest of you, here’s the story. A compressor is part of
a family of processors called ‘dynamics’ processors. These all respond
differently according to the level of the signal being fed to the
processor — and a preset designer can’t possibly know what level of
signal you’re sending to the processor! Furthermore, the greater the
difference between the quietest and loudest parts of the signal, the
greater its dynamic range, so a singer with good mic technique will
probably control their dynamic range to achieve a more consistent level
than a less experienced singer, and thus require less compression to
keep the level even in the mix. Again, the programmer of the compressor
preset can’t know how much compression your singer needs.

Level‑dependent
‘dynamics’ processors include compressors, limiters, expanders, gates
and de‑essers, and all have a threshold level that determines when
processing kicks in. Compressors, limiters and de‑essers are designed to
reduce the gain when the input signal exceeds the threshold level (they
act only on the loud bits). Gates and expanders reduce the level of the
signal when it falls below the threshold, usually to cut down the
amount of noise or spill in between the wanted portions of the audio
signal. In most cases, there will be a control called ‘threshold’, but
in some compressor designs the threshold is built into a single
‘Compression Amount’ control.

Compressors:
Typical compressors have lots of parameters that you can adjust, but if
you were to pick, say, a ‘Male Vocal’ compressor preset, the ratio,
attack and release times would probably be appropriate for vocals: all
you really need to adjust is the threshold, so that the gain‑reduction
meter shows the right amount of activity. (It’s still important to
listen, to check you’re getting the desired result!). As you lower the
threshold, more of the signal exceeds it, so more compression is
applied. Gentle compression, to even out the level, typically shows
a maximum gain reduction of about 5dB, but more assertive compression
(as you might use on rock or urban vocals) can require gain reduction of
around 10dB.

Compressors work by reducing the
level of the loudest parts of the signal, so the compressor output may
need to be ‘turned up’ to get the signal back up to its original peak
level. To this end, most compressors include an output gain control,
often called ‘make-up gain’. Some include an ‘automatic’ option for
this, but different plug-ins seem to handle this differently, and it’s
easy enough to set the make‑up gain yourself: adjust the level while
keeping an eye on the track or plug-in’s level meter, so that you end up
with a healthy signal level, while still leaving some headroom. Making
something louder will make it sound more impressive, but that’s no
different from raising the fader — so try to keep the compressed and
uncompressed signals at the same subjective level.

Spot
the difference! The compressor on the left has a threshold control,
which you adjust to suit the level of the source. The one on the right
has a fixed threshold: the input control raises the level of the source
above the threshold, so the compression kicks in, and the output control
compensates for any perceived changes in level.

Limiters: Rather than a variable threshold, plug‑in
limiters often have their threshold fixed at 0dBFS, and there’s an input
level control that can be used to boost the signal if you want to limit
the signal peaks for the purpose of maximising loudness. Essentially,
you only need to adjust the input level so that the limiter’s
gain-reduction meter shows a dB or two of activity on the loudest peaks.
If you want harder limiting, as you might on an individual drum, just
turn the gain up a little more; let your ears decide when enough is
enough.

Gates & Expanders: Like compressors,
gate and expander presets will have their attack and release times
tailored to specific applications, such as drums, vocals, guitar and so
on. Again, though, you’ll need to adjust the threshold to get them to
work correctly for your track. The simplest way to set them up is to set
the track to loop around a section that contains both wanted signal and
pauses; then, starting with the threshold at minimum, increase it until
the noise in the pauses just disappears. Don’t set the threshold any
higher than necessary to mute the noise, though, or you may start to
hear the wanted signal being affected, especially where the signal has
a wide dynamic range, or long, decaying tails. If you really want to
minimise the chance of adverse side-effects when tweaking a gate, reduce
the amount of attenuation (most gates, though not all, include
a control for this) so that instead of muting the pauses completely, you
simply reduce the noise to an acceptable level. Attenuating by between
10 and 20dB is often enough to keep the track sounding clean.

De‑essers:
De-essers vary in design, but most have a variable sensitivity control,
which essentially sets the threshold above which de‑essing takes place.
A de‑esser monitors the frequency band most likely to contain sibilant
‘S’ and ‘T’ sounds, and then applies gain reduction to that part of the
audio spectrum when loud sibilants are detected. Often there’s no
gain‑reduction meter, so you simply have to adjust the sensitivity
control by ear. What you’re after is a setting that reduces the ‘spitty’
character of those ‘S’ and ‘T’ sounds, but that doesn’t go so far as to
make the vocal sound lispy.

Some specialist
plug‑ins that reshape the envelope of a sound, or those that create
swept‑filter effects, also rely on a threshold setting to adjust how
they respond. The main exception is the SPL Transient Designer (and its
many imitators), which intelligently varies its own threshold according
to the input source, and works on material recorded at pretty much any
level.

Equalisation

How
much EQ boost and cut you apply depends entirely on the source material
and the track. To make use of EQ presets, it’s best to keep the
frequency bands fixed, but tweak the amount of cut and boost. Be careful
not to end up simply boosting all frequencies though!

When it comes to EQ, the preset designers will once
again have had to make assumptions about the source signal — so the
presets won’t be a perfect match for your track. A Male Voice EQ preset,
say, might not work in every case: for example, if there’s already
plenty of 2.5kHz attitude in the voice, you’re not going to want to
boost that band further! That said, the preset designer will at least
have carefully chosen the frequencies that are likely to have the
greatest impact on a typical male vocal — so usually the simplest way to
tweak such a preset is to focus on changing the gain setting for each
EQ band, without changing the frequency or Q. Gain or attenuation in
these bands should make plenty of difference, even with reasonably small
cut or boost settings.

Keep comparing the
result with the clean (bypassed) sound, as it’s very easy to fool your
ears into thinking louder and brighter means better! As a rule, use as
little EQ as you can get away with to do the job. Using EQ effectively
is a very necessary skill for any recording engineer, so also try to
wean yourself off presets as soon as possible. There are lots of
articles covering EQ on the SOS web site, for example at www.soundonsound.com/sos/dec08/articles/eq.htm.

After
experimenting with the gain settings, the best way to ease yourself
into parametric EQ is to experiment with varying the frequencies of the
various bands to see what audible effect that produces. When you feel
comfortable adjusting these two, move on to adjusting the Q or
bandwidth, which controls the width of the cut or boost region.

Delay

As
with compressors and most other dynamics processors, a gate (such as
the Sonalksis one pictured here) requires you to set the threshold to
make the preset settings work on your material.

There are three main adjustments you might want to
make to a delay plug‑in: how much, how long and how many repeats. If
you’re using a delay as an insert effect, the mix control determines how
much of the delay you hear relative to the dry sound. If the delay is
set up in a send/return loop, the mix control should be set to 100
percent wet. ‘Time’ determines the delay time (in milliseconds or in
musical measures), and ‘feedback’ controls how much of the delayed
signal is fed back to the input to produce further repeats. Many delays
also include a tempo‑sync feature, which is only relevant if you
recorded your song to the tempo grid, but it’s obviously helpful if
you’re programming and experimenting with different tempos.

Many
engineers record live bands without using the tempo grid, and though
this makes editing a little harder, it can free the music from the
‘tyranny’ of the click track, which, in turn, can help the song
‘breathe’ through natural small tempo changes. In such cases, you’ll
need to turn off tempo sync and set the delay time by ear.

Reverb

Reverb
comes in so many different flavours now, but back in the day, the plate
reverb was about the only game in town, and I find that it still sounds
right on almost anything. The main parameters that control the amount
of reverb are the mix setting and the decay time. As with delay, set the
mix to 100 percent wet if using in the send/return loop.

If
you’re confused by the reverb options available, try starting your
project with a vocal and a drum plate preset on two different effect
sends, and then adjust each by varying only the reverb decay time to
suit your track. You could set up more reverbs, but that’s a great place
to start.

Reverb is such a taste‑driven thing
that any judgement is purely subjective, but as a very general rule,
there’s a trend on modern records to use less obvious reverb treatments
than on those made in the ’70s and ’80s. A useful tip is to try mixing
the different reverbs you’ve set up (as described above), because the
chances are they’ll have different decay times — and that means that you
can achieve useful ‘in between’ combinations by blending them.

Enhancers & Distortion

It’s
no coincidence that the biggest dial, smack in the centre of this 112dB
Redline Reverb, controls decay time: that’s probably the first control
you should reach for when tweaking a reverb preset to suit your mix.

Typical harmonic enhancers synthesize new
high‑frequency harmonics to brighten a sound that has little or no
natural top end. If you choose a preset, the filter frequency that
determines what part of the audio spectrum will be enhanced will already
be set, so to make adjustments you need only vary the harmonics
‘amount’ control.

Distortion plug‑ins are
popular for applications ranging from adding a little warmth to a voice
or synth to making an electric guitar sound like a blender full of
barnacles! Chances are that your DAW will offer you several different
types, with presets based on each — so try them all, to get an idea of
their tonal differences, and then adjust the ‘drive’ setting to add more
or less distortion. Many types of distortion are also dependent on
signal level, so the lower your recording level, the higher the drive
setting is likely to need to be to get the same result.

Modulation

Modulation
effects include chorus, flanging, tremolo, vibrato, rotary‑speaker
effects, and so on. While rotary‑speaker plug‑ins tend to have fast,
slow and stop settings, the others are often more adjustable, and the
general rule is that if you speed up the effect, you should also reduce
the depth setting — especially when dealing with chorus and flanging. If
you simply turn up the speed of a chorus or flanger without also
reducing the depth, you’ll tend to end up with a nausea‑inducing warble —
but then I suppose that might be what you’re after, so feel free to
experiment! Some tremolo and panner plug‑ins have a tempo‑sync option,
which is worth trying if it isn’t already engaged when you open the
preset.

Channel Presets

Some
DAWs allow you to save all the plug‑ins used in a channel strip as
a kind of ‘combo’, or ‘channel preset’. These can be called up for use
on another channel, or in another project, and can save you lots of time
when working with similar material. Although they can be a great
starting point in a new project, do bear in mind that the key parameters
for each plug-in, such as a compressor’s threshold, will probably
require tweaking for every track to which the preset is applied.

The Future Of Presets?

Though
not directly related to this article, some plug‑ins, such as the Waves
Producer series and the Izotope Nectar Vocal Suite, make life easy by
hiding away the controls that you’re not likely to need to adjust. That
way, you only have to deal with the important controls — which is an
approach that’s a bit like the preset editing advice given here. I find
this a good halfway house between tweaking everything and simply calling
up presets, and because of the work that has gone into designing
them, the results can be extremely good. Maybe we’re moving towards an
era where the software treats us more like musicians and producers, and
less like engineers?

Friday, November 29, 2013

Delay forms the basis for
a wide range of effects that can transform your tracks from dull and
pedestrian to polished and professional.

Geoff Smith

If I had to pick a single desert island
effect, it would be delay. Why? Well, delay isn't only an effect in
itself; it's also one of the basic building blocks for many other
effects, including reverb, chorus and flanging — and that makes it
massively versatile. The aim of this article is to focus on the
practical applications of delays, by looking at some ways they're
typically used on vocals, guitars and synths. I'll be writing primarily
about the sound of different delay configurations, and to make things
easier to follow, I've created an illustrative series of audio examples
(see 'Audio Examples' box for details). Note that I've used exaggerated
effects levels in the examples, to highlight the sound of the delay.

First, let's run through a quick overview to get any
newbies up to speed. As the name suggests, the way in which delay
processors work is quite simple: the programme material passes through
a memory buffer and it is then recalled from the buffer a short time
later. We refer to this time difference as the delay time. (In an
analogue delay, you could think of the electronics or the tape loop
performing the same function as the memory in a digital delay.) Multiple
echoes or 'repeats' of the programme material are produced by feeding
a percentage of the delayed material back from the output of the delay
buffer into the input. We refer to this as feedback. You should be aware
when adjusting the feedback parameter that high settings can result in
the level of the processed signal increasing rapidly with each repeat,
so if your monitoring levels are high, it pays to be careful!

1. A simple delay

2. A simple delay with feedback

As with reverb, delay is most often applied as a send
effect, rather than as an insert. This approach not only conserves
processing power when applying the same delay to multiple sources, but
it allows you to treat the delayed part of the signal, separately from
the original, with extra processing such as EQ or distortion and this
allows you more creative freedom.

That's it in a nutshell, but the combination of those
few parameters and the flexibility afforded by using delays as send
effects open up a world of production effects and tricks. With the
basics explained, let's work through some useful techniques.

During the mix process, vocals are often treated with either reverb
or delay, or possibly a combination of both, so in the following section
I'll work through some typical vocal delay treatments. To get the most
from these, call up a vocal part of your own in your DAW and use that as
raw material. For my examples, I've taken a heavily 'tuned' vocal with
a tempo of 130bpm (audio examples 1 and 1b).

1. Create a send from your vocal track and on the send, call up
a delay plug-in. It doesn't need to be a sophisticated one; in fact, the
simpler the plug-in is, the easier this will be to follow. I'm working
in Logic Pro here, and I've used Logic's Tape Delay.

2. Set the delay time to a quarter note and the feedback to zero. If
you're using a delay that doesn't have tempo-sync options, you can still
get things in sync by remembering the following equation: ms = 60,000 ÷
bpm

Note that 'ms' is the quarter-note delay time in
milliseconds, 60,000 is the number of milliseconds in a minute, and
'bpm' is the tempo in beats per minute. From there, you can divide the
result as necessary to get eighth notes, sixteenth notes, and so on.

3. With the vocal playing, gradually raise the send amount and you'll
hear a single quarter-note echo that, when set quietly, can add
a useful sense of ambience to a vocal.

4. Next, gradually raise the amount of feedback until the delay fills
up the gaps between the vocal phrases, but doesn't cloud over the
original (audio example 2).

With the quarter-note delay in place, it's common to try to create
a unique 'space' for the delay in the mix, and this can often be
accomplished by applying EQ before or after the delay (it doesn't
normally matter which, though there will be subtle differences in, for
example, the response of a tape delay to a signal with the low end
present and the low end filtered away). As Robert Orton, discussing his
mix of Lady Gaga's 'Just Dance' in SOS March 2009 explained, what EQ
settings you require will vary from mix to mix: "Sometimes you want
quite a dark delay that's hidden behind the vocals just to give it more
body; at other times [there's] a word that clearly repeats, in which
case the delay has to sound up-front and clear”.

Many plug-in delays have in-built high- and low-pass
filters, but don't worry if yours doesn't, because you can use
a separate EQ — as in this example:

1. Find an EQ plug-in that includes high- and low-pass filters and place an instance of it after your delay.

2. Start by rolling off the high frequencies of the delayed sound,
using a low-pass filter. This will soften any transients, and help to
push the delay back in the mix behind the main vocal sound. When used at
low levels, this creates a really subtle sense of ambience.

3. Now bypass the low-pass filter and try high-pass filtering the
delay, to remove some of its bottom end. This can help to stop the delay
clouding up the low end of your mix.

4. Finally, combine the high-and low-pass filters to create
a 'telephone' EQ. This effect (see image 2) is much less obvious when
used on the delay signal than on the source vocal track, and it's
a tactic that can reduce the amount of space taken up by the delay,
leaving space for other elements. It also helps to make the delay more
distinct from the original vocal (audio example 3).

You can find another example of this last effect in Peter Mokran's
Mix of the Pussy Cat Dolls track 'Jai Ho!', where he inserted the
telephonic EQ before the delay (for more details, see the interview in
SOS August 2009's Inside Track feature, /sos/aug09/articles/it_0809.htm).

One reason for using a delay instead of a reverb for
these treatments is that quarter- or eighth-note delays are heard
distinctly from the original vocal, whereas with a reverb, the early
reflections can easily combine with the original vocal part and thus
change its perceived timbre. This means that, for example, adding
a bright quarter-note delay doesn't brighten the original vocal in the
way that adding a bright reverb might do.

3. 'Telephone EQ' can be applied before or after the delay effect to differentiate the echo from the original vocal.

Another way to make the delayed signal stand apart from
the original vocal is to run the delay through an amp and/or speaker
simulation plug-in. The plug-in will not only provide a unique
equalisation curve, but the sound of the speaker and the distortion from
the amp will add a compression effect that can help to even out the
level between repeats. This adds a greater sense of sustain to the
delayed signal, which you can hear in audio example 4. It's worth
experimenting with different amp models and cabinets, as you'll find
plenty of different timbres. There really are no hard and fast rules as
to what works best here.

Now let's set up a second send, this time with a eighth-note delay.
Again, there are plenty of plug-ins you could use, and I'm going to
choose Soundtoys Echoboy, which is a really versatile plug-in. It has
low- and high-pass filters and, in the Style Edit section, further
equalisation and output-modelling options. This allows you to do all of
the tonal shaping we've just discussed inside a single plug-in, and this
means that you can quickly compare radically different delay,
equalisation and distortion settings simply by changing presets. Echoboy
also has a Decay EQ section, which is an EQ that's placed in the
delay's feedback loop, and thus allows you to make each successive
repeat duller or brighter, making the delay evolve in timbre with each
repeat (audio example 5). In simpler terms, this feature controls how
the tone of the echo changes over time.

There are plenty of other plug-ins that offer this sort
of feature, and it's well worth having one in your mix toolkit, as it's
often tricky setting up an EQ'ed feedback signal using separate delay
and EQ plug-ins, partly because not all DAWs have sufficiently flexible
routing.

With conventional delay setups, it can sometimes be difficult to find
a constant send level that allows the delay to be sufficiently audible
when the singer finishes a word, but doesn't cloud the vocal when he or
she is singing. To counter this problem, you could use automation to
ride either the send or return level of the delay on the offending
passages (better to automate the send if you're sharing the delay with
other sources). This has the advantage of giving you absolute control
over the level of the delay, but it can be fiddly, particularly if you
need to go back and change this automation as your mix evolves.

A less fussy means of achieving a similar result is to
set up a compressor to duck the delay when the vocal is present. To set
this up, you'll need a compressor and DAW that allow you to use an
external side-chain input. Here's how to do it:

1. Add the compressor after the delay — and any subsequent processors in the send chain — on your send channel (picture 5).

2. Next, go to the compressor's side-chain input and set it so that
the compressor is triggered by the lead vocal track. How you do this
varies from DAW to DAW, so if you're unsure, check the manual.

3. With the track playing, lower the threshold and raise the ratio to
apply the appropriate amount of gain reduction to the delayed signal
(audio examples 6 and 7).

This is a common but effective trick, which has been used on plenty
of hit records. By way of example, Marcella Araica used this on
Timbaland's 'The Way I Are': listen to that track, and you'll hear the
delay become prominent after the last lines of the verses.

5.
The setup in Logic for creating a ducking delay. I've used Logic's
Compressor plug-in to provide the ducking but I could just as easily
have used the included Noise Gate or Ducker plug-ins, shown bypassed.

When you're mixing, one of the advantages of using a simple
quarter-note delay on a vocal is that, compared to reverb, it takes up
such a small amount of space in a mix, and this is even more the case
when you use a mono delay. By combining a single quarter-note mono delay
with a lead vocal that's also mono, you can preserve the positional
integrity of that source. This can be helpful in busy arrangements,
where space is at a premium.

Mixing isn't always about saving space for other
elements, though: in a spacious mix, you might actually want the vocal
effects to take up more room, and in this scenario it's a good idea to
look at creating a sound using both your delay and reverb plug-ins.
Running a delay into a stereo reverb will spread the delay out across
the stereo field, making it seem broader (audio example 8a). One trick
that's well worth exploring is running your delay through an
early-reflection patch from a reverb, as this will 'stereoize' the delay
without it sounding so obviously effected (audio example 8e).

You can also use both your mono delay and stereo reverb
to deliberately create contrast between song sections. For example, you
could use a quarter-note delay on a mono verse vocal to add a sense of
ambience while keeping things dead centre, but follow this in the chorus
by moving to ultra-wide stereo vocals, with stereo delay and reverb
effects. It's this sort of contrast that's so essential in keeping the
listener's ear engaged. When processing a lead vocal, then, consider in
all of the different song sections whether you want both the vocal track
and its send delay and reverb effects to sound more mono or more
stereo.

A delay can also be used to change the character of
a reverb, by acting as an ultra-configurable pre-delay. If you listen to
audio example 8c, you'll hear that the vocal is treated with a small
amount of hall reverb. Then listen to example 8d, where I've taken the
same reverb but inserted an eighth-note delay (with feedback set to
zero) before the hall reverb: the delayed reverb seems louder and more
obvious in the mix. Even though many reverb plug-ins have a pre-delay
control, they're often not tempo-syncable, and that's why I prefer to
use a dedicated delay plug-in to provide the pre-delay: most delay
plug-ins offer tempo-sync'ed delay times at the click of a button. Thus,
I'm able to quickly compare say a sixteenth-note or eighth-note
pre-delay.

Ping-pong delay is a type of dual delay where the first echo appears
in the 'ping' channel (usually the left), delayed by the ping amount,
and the second appears in the opposite 'pong' channel, delayed by the
ping time plus the pong time. For example, if you set the ping to 200ms
and the pong to 400ms, you'd first hear the ping 200ms after the
programme material out of the left channel, and the Pong 600ms after the
programme material out of the right channel. This process will then
repeat, assuming the feedback values are higher than zero. In audio
example 9, Echoboy is set to a ping-pong delay with the Transmitter
output style, to give a telephonic effect.

On his mix of U2's 'No Line On The Horizon', Declan
Gaffney used the SoundToys Echoboy plug-in to add a subtle amount of
ambience to the lead vocal using a ping-pong setting: "The echo is just
a kind of warm ping-pong sound. There's no reverb on the track, there's
not even any feedback on the delay, it's all about the dry sound with
a little bit of space on the side, provided by the delay… I'm not a huge
fan of reverb anyway; it's better to do the same thing with delays.”
(SOS June 2009, /sos/jun09/articles/itu2.htm). Audio example 9b uses a similar setting.

I'm not sure if there's an official definition of 'slapback delay' as
an artificial effect, but it is generally used to describe a single
echo of 60-180ms that creates a sort of thickening effect. Famous fans
of slapback on their voice include John Lennon and Elvis Presley. 'Slap
delay' like this is also a standard treatment for hip-hop vocals
because, as Jaycen Joshua put it in SOS August 2010, "reverb is the kiss
of death on rap vocals” (/sos/aug10/articles/it-0810.htm). Audio example 10a demonstrates how slapback delay can be used to enliven a vocal part. The delay here is set to 80ms.

Mix
engineer Jaycen Joshua, who explains that "reverb is the kiss of death
on rap vocals,” but still uses delay to enliven the sound.Slapback
delays were originally created with tape machines, so you may want to
try rolling off a little of the top end and adding a small amount of
modulation of the delay time to approximate some of the inconsistencies
of tape. It's also fun to try a stereo slapback delay using a dual delay
with slightly different delay times for each side, as this will create
a wider stereo image. Audio example 10b is a slapback delay with the
left side set to 80ms and the right side set to 120ms: listen out for
the extra stereo width this gives the vocal.

I'll mention one last slap-related trick before we move
on: take a short slapback delay and then gradually increase the
feedback. Because of the relatively short delay time, this will begin to
sound reminiscent of a spring reverb, such as you often find in guitar
amps.

The German scientist Helmut Haas wrote that when two identical
signals, each played through a separate speaker, are delayed by anything
from 1-30ms, a sense of a broadening of the primary sound source is
heard, but without there being a perceptible echo. This effect, often
referred to as the 'Haas effect', can be created using delay plug-ins or
track offsets, and can be used to add stereo width to a vocal part. It
also serves as a reasonable foundation for creating fake double-tracking
effects.

Let's look at some settings. If you take a delay
plug-in and increase the right channel's delay time to 10ms more than
the left, you should notice that the delay creates a stereo effect
rather than a perceptible echo: the delay simply broadens the vocal,
adding stereo width (see audio examples 11a and 11b).

Using a plug-in like Logic's Sample Delay can create extra stereo width, but often at the expense of mono compatibility.

Jaycen Joshua used this type of processing to create
a stereo effect for the crash cymbal on Justin Bieber's 'Baby': "There's
a medium delay on the crash,” he remarks, "because it was originally
a mono track, and I wanted it in stereo. So I set the delay to 21ms and
mixed the original 100 percent to the left, and the delay to the right….
With 21ms, you get enough separation between left and right and it's
a bit dramatic and not so phasey.... The sound becomes like a drummer
hitting two cymbals, left and right, at the same time.”

Unfortunately, though, this approach can create
problematic side-effects. First, simply delaying one side in relation to
the other will result in unpleasant comb filtering when the left and
right channels are summed to mono. Example 11c shows the effect of
making the delayed vocal mono. Contrast that with example 11a and you'll
hear why this sort of processing is risky if you want to ensure mono
compatibility! A second problem with this treatment is that you can
perceive that the undelayed side as louder than the delayed side,
causing potential balance problems.

To take advantage of the widening effect that Haas
delays offer, while deftly dodging the mono-compatibility trap, there
are some useful tricks. One of the most common widening treatments is
created using two short delays, with a small amount of pitch-shifting
applied to either side. To achieve this effect yourself:

1. Set a mono delay to 11ms and with zero feedback, then pan it hard right.

2. Then, add a pitch-shift plug-in to take it up seven cents. Repeat
this with a second delay, but this time pan it hard left, and
pitch-shift it down by seven cents.

For a demonstration of this effect, see audio examples 11d and 11e.
To create these, I've used Logic's Delay Designer plug-in (picture 7),
as it allows you to set up multiple delays and pitch-shifting all in the
same plug-in, but it's easy enough to do by combining delays and
pitch-shifters separately. If you want to hear the effect used in
context, there are countless commercial tracks I could offer by way of
example, one of which is the Arcade Fires album The Suburbs. Producer
Craig Silvery explained in SOS November 2010 that he used "two short
delays from an AMS with a little pitch-shift, one up and one down, which
thickened the vocals like a doubler” (/sos/nov10/articles/it-1110.htm). There are also plenty of examples of this tactic being used to good effect in our regular Mix Rescue articles.

A nice variation on this effect is provided by Waves'
Doubler plug-in, which provides the same widening effect but with a less
obviously effected sound. In the words of Grammy-winning mix engineer
Dave Pensado: "Doubler has four delays that also help to make the vocal
sound bigger, wider and more powerful” (SOS Jan 2007, /sos/jan07/articles/insidetrack_0107.htm). Audio example 11f provides an example of this setting.

Very short ping-pong delays can also make excellent
wideners, as you can hear if you listen to audio example 11g, which
shows a 30ms ping-pong delay built with NI Reaktor, where the output of
the ping delay is polarity inverted before going into the pong delay.
This technique is great for transparent widening and has excellent mono
compatibility.

A multi-tap delay is a delay line where multiple 'taps' or outputs
are taken from a delay buffer at different points, and the taps are then
summed with the original. Multi-tap delays are great for creating
rhythmic delay patterns, but they can also be used to create sound
fields of such density that they start to take on some of the qualities
we'd more usually associate with reverb. Favourite plug-ins for the job
include Waves Supertap, PSP Audioware's PSP608 and Echoboy, using its
Pattern mode, but there are many more available, and you might even have
suitable toys bundled with your DAW (I often use Logic's Delay
Designer, for example).

It can be great fun using a multi-tap delay to design
reverb-like effects that, while they might not compete with a proper
reverb algorithm in terms of realism, can produce some wonderful,
unique-sounding results. A simple method for creating a reverb-like
setting is to take a multi-tap delay and create a series of delay taps
starting at 30ms and increasing in time. With the preset shown in
Picture 8, I've gradually increased the delay tap times in Logic's Delay
Designer at random up to the last taps, which are in time with the
sequencer tempo, the last tap being a quarter-note delay. To build on
this starting point, experiment with panning successive delay taps left
and right, filtering the delay taps so that each tap becomes duller, and
changing each tap's volume. In audio example 12, the taps swell in
volume toward the middle of the delays and then fade out again in the
later taps.

8. Consecutive delay taps panned left and then right to create an effect similar to reverb.

9. In this screen, the delay taps are being increasingly high- and low-pass filtered the longer they get.

Dave Pensado describes how he used the Waves Supertap
plug-in on the lead vocal for the track 'Beep' by the Pussycat Dolls:
"The delays on Supertap are all very short, and what it also allowed me
to do is spread the vocal wide across the stereo spectrum. In other
words, instead of occupying a small spot in the middle of the mix,
I could fill the whole spectrum between the speakers. The 149, 298 and
587 ms are sixteenth, eighth and half-note delays, and they spread and
get louder from left to right." If you have Supertap in your plug-in
collection, you can find a similar setting to the one used for 'Beep' in
the preset menu under Dave Pensado/Pensado Tap Vocal — and if not, you
can still hear the effect in audio example 13.

10. Waves Supertap plug-in, a good example of a very tweakable multi-tap delay.

Of course, you don't have to use just one delay effect on a sound.
It's perfectly possible to set up a range of different delays on your
aux send channels, and then experiment with automating different delays
on different words. It's this level of sophistication you might need to
reach for if you want to compete with contemporary pop tracks. For
example, in SOS March 2009 (/sos/mar09/articles/it_0309.htm)

Robert Orton, who used extensive automation of delays on Lady Gaga's vocal for the smash hit 'Let's Dance'. Robert
Orton described how he automated the delays on Lady Gaga's hit 'Just
Dance': "When soloing the vocals, I added half-, quarter- and
eighth-note delays, and I think there's also a dotted eighth-note delay,
all using the Sound Toys Echoboy... The eighth-note delay is panned to
the right, and comes in the choruses and some words in the verses. The
send is automated. The half-note delay is panned to the left, and
captures certain words; for instance, in the chorus each time the word
'dance' occurs at the end of a line. The quarter-note delay is also
panned to the right, and is automated to happen on certain words. All
the delays catch words differently, to keep it interesting. They're also
set to different styles on the Echoboy — TubeTape, Analogue, etc — to
get different textures.”

All of the techniques from the vocal section can also be applied to
other instruments, including the guitar. For example, quarter- and
eighth-note delays can be used for ambience and delay/pitch wideners to
add stereo width. But there are also plenty of effects that work
particularly well on guitar.

In this section, then, let's work through an example of
using delay on guitar, to show how it can transform even the simplest
of parts. We'll try to emulate one of the most famous and
distinctive-sounding delayed guitar sounds: that of U2's The Edge, which
you can hear on many records, including The Joshua Tree. The delay
effect that's most often associated with the Edge has a delay time of
three-sixteenths or a dotted eighth-note of the song tempo.

U2
guitarist The Edge forged his signature sound using delay, which you
can hear on classic albums such as The Joshua Tree. Find out how to
mimic this sound here!

1. It pays to start with a really simple, repetitive guitar part, so
for my audio examples I've created a simple eighth-note guitar line
(audio example 14a).

2. Now, place a 3/16 note delay against this, to create a much more
complex pattern (audio example 14b). For the example, I used a Korg
SDD2000 hardware delay unit (from the same time period as The Joshua
Tree), but you can get good results with a plug-in delay too.

3. Next, experiment with using a very slow LFO to modulate the delay
time. Used subtly, this creates a pleasant amount of movement in the
delay. Some plug-ins include an LFO, but if yours doesn't, you can
create a similar effect by routing a separate MIDI LFO plug-in to
control the delay-time parameter, or draw the modulation in using
automation.

4. Finally, try experimenting with panning the delay into a different
position from the dry signal. To demonstrate this, I created audio
example 14c, in which I've left the guitar in the centre and panned the
delay hard right. More interesting stereo delays can often be achieved
by choosing two different delay timbres, with one panned hard left and
the other hard right, and with slightly different modulation rates (see
audio example 14d).

Heavily
compressing a synth sound after the addition of delay and reverb can
give the part an interesting sense of movement, as the effects are
effectively ducked when the synth is played. Here, I've used Logic's own
plug-ins to apply this effect to a simple plucked-string patch to
create a much more complex, and richer-sounding result.

Up until now, we've looked mainly at using delays as send effects,
but now let's consider an example of where you might use them as
inserts. When using a synth, there are some advantages to processing the
delayed signal along with the dry, and to understand why this is the
case, let's work through another example:

1. Take a simple plucked synth and program a repetitive pattern based around eighth notes (audio example 15a).

2. Now add a stereo delay and slightly offset the right and left
delay times to increase the stereo width of the effect (audio example
15b).

3. This sounds quite boring at the moment, so now let's make it more
interesting. Add a small amount of a medium-sized reverb (audio example
15c).

4. Next, insert a compressor after the delay and reverb. Set this so
that it's applying about 10dB of gain reduction, and adjust the attack
and release time so that the compressor clamps down on the original
pluck and then releases from compression in time for the next delay.

Hopefully, what you can hear is a transformation from a boring,
predictable patch to one with some movement. Notice that the delay and
reverb elements are also brought forward in the mix (audio example 15d).
Essentially, what's happening is that the compressor applies 10db of
gain reduction to the whole effects chain whenever the plucked synth
plays, effectively ducking the delay and reverb by 10dB every time the
synth is played. This creates a really satisfying pumping effect and
gives the part a wonderful sense of rhythmic movement that simply
increasing the effect level couldn't match. There are ways to achieve
this with sends, using side-chain compression (remember the ducked
delay?) or sending both the source and delay tracks to a bus, but it's
much simpler to use delay as an insert in this case!

We don't need to stop here, though: we can create more
rhythmic complexity in the same part by stealing ideas from the Edge, so
adjust the left delay time of your stereo delay to 3/16th notes while
keeping the right delay to 1/8th note (audio example 15e.) Again, listen
to how the compressor effectively ducks the effect each time the synth
plays.

Now we'll use delay to take the synth part to
a climactic peak. When you turn up the feedback on a delay, it will
begin to self oscillate, and the delay will get louder and louder. Dance
musicians have managed to take advantage of the excitement this effect
creates, but keep it under control, by using a limiter to pin down the
delay's output level, so that high feedback amounts that would normally
result in ear-bleeding pain are kept manageable.

1. Use the delay from the previous example and then insert a limiter
at the end of the effects chain. Set the limiter so that the maximum
output is something bearable (mine is set to -7.5dB), and set the
threshold so that you can see a small amount of gain reduction on the
meters under normal circumstances.

2. Now, slowly bring up the feedback of the delay. As the delay moves
into self oscillation, you should find that the limiter pins the delay
to a bearable level instead of allowing it to increase in volume like
a crazed animal.

This article presents only a few ideas for using delay in your
productions, and there are whole, beautiful worlds that are swamped in
delay effects and waiting to be discovered. Have fun exploring them! .

Article Preview :: Clean Up Your Acts

Technique : Effects / Processing

Restoration software such as Izotope's RX2 can breathe new life into damaged audio — with the right moves from the user!

Mike Thornton

Izotope's RX2 software makes available
some of the most powerful restoration tools around, at an affordable
price, and as a result has become a very popular package. In this
article, I'm going to share some power-user hints and tips that will
help you get the best from it.

If you're new to RX, a good place to start would be checking out Sound On Sound's original review in July 2008 (/sos/jul08/articles/izotoperx.htm).
Since that review was published, Izotope have released a new and
significantly expanded version called RX Advanced, in addition to the
basic RX. RX Advanced has a number of extra modules, and some of the
modules that appear in both versions have extra features in the Advanced
release. Both variants can be run as stand-alone applications or as
plug-ins for your favourite Mac OS or Windows DAW: Audio Units, VST,
RTAS and AAX Native plug-in formats are supported. However, running RX
as a plug-in means that its processes have to operate in real time. This
makes some features unavailable and limits the effectiveness of others,
so I tend to export files that need processing to the stand-alone
version of RX and re-import them into my DAW once processed. (For this
reason, my number one feature request would be for Izotope to improve
the links between DAWs and the stand-alone application, perhaps in the
same way as Synchro Arts have done with Revoice Pro.)

The basic version of RX has five main restoration
modules. Declip is for repairing clipped and distorted audio, while the
Declick & Decrackle module is intended for restoring recordings from
vinyl records, although it is also good for dealing with digital
clicks. Remove Hum can eliminate low-frequency noise such as mains hum,
along with up to seven harmonics. Denoise removes broadband noise that
is relatively static in profile; it is effective both on electrical
noise such as hiss, and acoustic noise such as air-conditioning.
Finally, Spectral Repair is designed to remove occasional random sounds
that have interrupted a recording, whether these come from the
instrument being recorded or from external sources. Beeps, car horns,
mic pops and mouth clicks are all grist to its mill.

Supplementing these are a selection of 'fix it' modules
such as Gain & Fades and Channel Ops, which can help with all kinds
of routing and phase-related problems. There is also a Spectrum
Analyser module to help track down exactly where the problems are. In RX
Advanced, there are more 'fix it' modules you wouldn't necessarily
expect to find in a restoration software package, such as Resample for
downsampling audio files, Dither for reducing the word length of audio
files, and pitch-shifting and time correction using Izotope's Radius
technology. The Advanced version also operates as a VST and AU plug-in
host in stand-alone mode, and boasts an intriguing Deconstruct module,
where you can separate and adjust the tonal (pitched) and broadband
(unpitched) elements of an audio file.

To illustrate some of the techniques involved in
getting the best from the various RX modules, I'll work through some
examples of audio files that have needed some work. I will describe how
things happen in the stand alone application, but most of what I am
covering could be undertaken in the plug-ins within the limits of
real-time processing.

Digital audio is not forgiving when it comes to peak distortion: if
your signal exceeds 0dBFS, you will experience clipping, with anharmonic
distortion that is usually very obvious and unpleasant.

My first screenshot

1: The stand-alone RX Advanced. A stereo audio file is loaded that has severe clipping distortion.
shows an audio file where the tops of the waveform have been chopped
off, and the challenge is to restore this audio to its former glory
using the Declip module. Either adjust the Clipping threshold by eye so
that either the red lines on the audio waveform are below the
clipped-off peaks, or click the Compute button in the module window and
adjust the Clipping threshold control until the red line is just before
the white line in the module window display. Next, adjust the Makeup
gain. This is actually an attenuation control, and it is needed because
once RX has reconstructed the missing peaks on the audio, its new peak
level will be around 6dB higher — hence the default setting of -6dB. If
you are already close to 0dBFS, you might want to consider reducing this
further.

2:
RX's Declip module. 'Makeup gain' is a misnomer, as it's actually
attenuation that is required following the de-clipping process.

Thursday, November 28, 2013

Performance Synthesizer

Reviews : Keyboard

The Jupiter 8 looms large in
synthesizer history, and any synth bearing the name has a lot to live up
to. Is the Jupiter 80 destined for the same legendary status? Find out
in our world‑exclusive review...

Gordon Reid

I have friends who have been waiting
nearly three decades for a successor to the Roland Jupiter 8. Their
hearts went all a‑flutter when 1991's JD800 was announced but, while
this is now a minor classic in its own right, it wasn't what they had
envisaged. They went through the same set of emotions when the JP8000
appeared in 1997, only to be disappointed again. But now there's a synth
that says to the world, "Let there be no confusion; I am a Roland
Jupiter”. Launched amid a flurry of speculation, praise and diatribes in
equal measure from people who had never been within 100 miles of one,
let alone heard one, it's the Jupiter 80.

Physically, it's somewhat larger and heavier than
Roland's most recent and now discontinued 76‑note workstation, the
Fantom G7. Its colourful control panel is reminiscent of a Jupiter 8,
but only in a superficial way, and it's clear even before switching it
on that most of the action is going to take place on the 800 x 480-pixel
touchscreen that dominates its control panel. The touchscreen is good
news; I've lost track of the number of times I've poked a Fantom's
display in the expectation that something will happen.

The Jupiter 80 generates its sounds using the
Supernatural technology first heard on the ARX boards introduced for the
Fantom G series, married to a significantly cut‑down version of the APS
(Articulative Phrase Synthesis) technology found in the V‑Synth GT.
However, despite the justified clamour from Fantom owners, there are
only three ARX boards (one each for drums, electric pianos and brass),
and the set of polyphonic APS sounds in the Jupiter 80 does not overlap
fully with the APS sources and Phrase Models in the GT, so it's clear
from the outset that the new synth is not simply a mélange of existing
engines presented in a colourful new box.

What's even more apparent is that the Jupiter 80 is not
based on any conventional synth architecture, because it eschews the
conventional patch and performance structures that have dominated synth
architectures for the past 20 years or so. The lowest level (or so
Roland claims) is the 'Tone', and there are two distinct Tone
generators: Supernatural Acoustic (which, confusingly, also contains the
APS sounds) and Supernatural Synth. The next level up is the 'Live
Set', which can comprise up to four Tones in 'Layers'. The top level is
the 'Registration', which comprises four 'Parts': a single Tone in the
Perc Part, a Live Set in the Lower Part, another Live Set in the Upper
Part, and another Tone in the Solo Part. Confused? I'm not surprised; so
was I. But I have to admit that, by the end of the review, I found it
simple to use, if rather unusual.

Resemblance
to its ancestor aside, the Jupiter 80's large, colourful buttons and
simple control panel make it especially useful for live performance.

Let's start at the lowest level and look at the Supernatural Acoustic
Tones. There are 117 of these, divided into categories such as pianos,
basses, strings, guitars, and so on. A closer look shows that some
sounds are presented in two versions, with the second prefixed by the
letters 'APS'. Consequently, you have (for example) 0062:Oboe and
0104:APS Oboe, both of which sound like oboes but nevertheless sound
quite different from one another. There's nothing to worry about here,
although Roland seem to have got themselves into a bit of a semantic
tangle when combining the Supernatural and APS technologies, because the
literature also talks about something called Behavior Modeling
Technology (the company's spellings, not mine) which, like APS, also
claims to emulate the behaviour of a given instrument when you play its
physical model. Are APS and BMT components of Supernatural Acoustic, or
is Supernatural Acoustic the initial sound generator and are APS and BMT
independent performance modifiers? Or is APS a component of BMT? Who
shot JFK? Damned if I know!

So let's turn to Supernatural Synth, where things
should be simpler. Except that they're not. Indeed, I found the Synth
mightily confusing until I realised that Roland's claim that the Tone is
the lowest level of sound creation is wrong. I initially approached
Supernatural Synth on that basis, but I got myself into a tangle because
I wasn't differentiating between the controls that affect
a Supernatural Synth Tone as a whole, and those that programme the three
miniature synthesizers ('Partials') that comprise it.

A Partial is a powerful synthesizer in its own right.
Its oscillator appears to offer eight waveforms, but the six
analogue‑type waves each have three variants, and pulse width and PWM
are programmable where appropriate. The depth of the Super Saw (the
seventh option) is also programmable, while the eighth option allows you
to select any one of 380 PCMs that include many of the underlying
waveforms from earlier generations of Roland's digital synths. There's
also a dedicated AD pitch envelope, a ring modulator between Partials 1
and 2, and a waveshaper that can act upon any of the resulting sounds,
whether Virtual Analaogue or PCM digital. Similar flexibility is
apparent when you turn to the multimode (low-pass, high‑pass, band-pass
and peaking) filter with its 12dB/oct and 24dB/oct slopes, and to the
amplifier. There are even two LFOs, one for conventional duties, and
a second dedicated to the modulation joystick. This is all good stuff,
and a polysynth built on three of these (ie. a single Supernatural Synth
Tone) would be a very powerful instrument in its own right.

Largely
eschewing the knobs and sliders of its forebear, the Jupiter 80's synth
engine is accessed mostly via the 800 x 480-pixel touchscreen.

Moving up a level, you can select any four Tones — whether generated
by Supernatural Acoustic or Supernatural Synth — to insert into the four
Layers in a Live Set. Strangely, this is also the level at which Tone
editing (as opposed to Synth Partial editing) takes place, so before
I can tell you about the facilities provided by the Live Set itself,
I need to tell you about what it lets you do to the Tones that comprise
it.

Let's say that you want to layer a couple of Tones —
one generated by Supernatural Acoustic and the other generated by
Supernatural Synth — within a Live Set. To do so, you place the first
in, say, Layer 1, and the second in, say, Layer 2 of the Set. (You have
to do this when the Live Set is inserted within either the Upper or
Lower Part of a Registration, but let's ignore that for the moment.) Now
let's say that you want to edit the first of these Tones. Punching the
appropriate Edit button reveals a handful of parameters that are
relevant to the instrument in question, so, for example, the piano model
offers control over string resonance, key-off resonance, hammer noise,
stereo width, nuance and tone character, while the flutes offer noise
level, growl sensitivity and 'variation', while the electric pianos
offer just a single parameter: key-off noise. In contrast, the most
complex of the models, the electric organ, is based on a Hammond
generator with all nine drawbars, percussion, keyclick, leakage, and
even subtleties such as the unusual behaviour of the 1‑foot drawbar
correctly implemented. Frustratingly, there's no user memory for edited
Acoustic sounds, so you can only save the modified Tone within the Live
Set in which you edited it. I can see that this saves memory but if, for
example, I want to create and store a range of Honky Tonk pianos,
I can't do so except by using up Layers and Live Sets. Perhaps the
reasoning was that there are so few parameters in a Supernatural
Acoustic sound that any edits could be recreated elsewhere without too
much hassle.

Moving across to Supernatural Synth, inserting a Tone
into a Live Set provides what looks like a complete set of synthesis
parameters in addition to those found at the Partial level. However,
these are not absolute, they are modifiers that override some of the
values in the Partials (such as switching the filter type to a new
setting) or provide offsets that affect the Partials' values. This is
a weird architecture. The system is, in effect, treating the Partials as
an 'expert' level, allowing novice users to regard Synth Tones as
immutable building blocks and to tweak them into shape using the
controls provided at the Live Set level. However, the advantage of this
is that a Synth Tone moulded into a new shape in one Live Set is not
affected when inserted into another, which is not a trivial benefit.

Once you have inserted and, if desired, edited the
Tones within a Live Set, you can apply effects and other facilities to
them. On the surface, the effects section within a Live Set looks good,
with four assignable MFXs (multi‑effects) offering 76 effect types, plus
a global reverb. Unfortunately, the routing of the MFXs is fixed.

While
you can determine the level of the signal sent from each Layer to each
MFX, to the reverb, and to the outside world, the four MFXs are
permanently arranged in parallel, so there is no way to pass signal from
one to the next. This means that you cannot send sounds sequentially
through them to create (for example) an organ's effects path of chorus,
reverb, overdrive and rotary speaker, or through common guitar effects
paths such as compression, overdrive, EQ, chorus, and delay. I discussed
this with Roland, whose engineer's response was, "we could make the
routing more flexible, but this might have ramifications in other areas,
so we should consider this carefully”. In other words: don't hold your
breath. Notwithstanding this, the quality of the effects themselves is
up to Roland's usual high standards, which is hardly surprisingly since
many of them are augmented versions of the Fantom G effects, enhanced by
the welcome addition of a three‑band parametric EQ.

A Live Set also offers a variation on the concept of
morphing from one patch to another. Called 'Tone Blending', this allows
you to move from one state to another, with multiple parameters such as
level, filter cutoff, resonance and effect send levels being affected
simultaneously by a single knob or the onboard D‑Beam controller. You
can define the start and end points of the transition for each of the
Layers in the Live Set, which makes it possible to do things as simple
as introducing a sound into an existing mix, as interesting as morphing
from an Acoustic sound to a Synth sound and back again, or as
experimental as turning civilised patches into over‑effected sonic
mayhem. Then, if you stumble across something that you like anywhere
within the blending range, you can save this as another Live Set, which
is an innovative way to generate new sounds.

The
Jupiter 80's rear panel hosts all the connections you'd expect to see,
and a couple more besides. A full list can be found in the 'Abridged
Specification' box.

We now come to the top level, the Registration. There are 32 of these
(A‑1 to D‑8) in a 'Set', and eight Sets (called [01] to [08]), giving
a total of 256 Registrations, each of which comprises the aforementioned
four Parts — Perc, Lower, Upper and Solo — that can be played in
isolation, layered in a variety of ways, or split into a maximum of four
regions across the keyboard. If you want to change the whole setup at
the touch of a button, this is the level at which it's done, using the
buttons under the keyboard.

Given that the Solo Part comprises just a single Tone,
you simply insert the one that you want, determine the keyboard range
over which it will play, and set up things such as its level, pitch and
pan. The modifying parameters aren't as extensive as those provided by
a Live Set, and the Solo Part's effects structure — which offers just
compression, EQ and delay in series — is quite different from a Live
Set's, so a given Tone can sound quite different when inserted and
played here.

If anything, the Perc Part is the weirdest of them all.
With its Manual Percussion option selected, this sets aside the bottom
15 notes of the keyboard for a selection of percussion sounds. There are
eight sound sets provided but, given the limited number of notes
available, these are not laid out in conventional GM fashion.
Alternatively, you can select the Drums/SFX option, which gives you
access to 16 drum kits, either across the entire keyboard or limited to
the region defined by the Lower Part. You can also insert any
Supernatural Tone here, again accessible across the whole keyboard if no
splits are on, or in the same range as the Lower Part. This can be very
useful, although you have to remember that the Perc Part's editing and
effects have the same structure as the Solo Part's.

With more than a nod toward live performance, the
Jupiter 80 offers two additional tools at the Registration level that
can be applied to the Live Sets inserted into the Upper and Lower Parts.
The first is a powerful arpeggiator. This can generate traditional
patterns, with the usual parameters, such as octave range, rate and
shuffle, but when you start to experiment with its Styles, Variations
and Motifs, you'll find that it offers many more possibilities,
encompassing everything from simple patterns to guitar licks and strums,
walking bass patterns and more. You can even create up to 128 new
'User' styles by importing and saving Standard MIDI Files of up to 500
notes. The second tool is called Harmony Intelligence, and this adds
a harmony to the topmost note that you play in the Upper Part,
calculated from the notes that you play in the Lower. There are 17 types
of 'intelligence', and these determine the nature of the harmonies that
are generated. Names such as Big Band, Strings, Hymn, Country and
Gospel tell you exactly what Roland's engineers had in mind, but while
these would be appropriate for a domestic instrument, they seem a touch
incongruous here.

The Jupiter 80's MIDI capabilities are as extensive as
you would expect, with independent input and output channels for each of
the Parts, a separate control channel to change Registrations, MIDI
sync, extensive MIDI CC capabilities, and the ability to transmit
parameter changes as SysEx data. There's also an extensive menu for
controlling external sounds, and this allows you to set up things such
as velocity ranges and key zones for each of the 16 channels. In
addition to this, the Jupiter 80 offers Roland's proprietary V‑Link
protocol and is compatible with the new MIDI Visual Control
specification, both of which allow players to control still images and
video clips using MIDI note numbers and CCs. I would love to have access
to this technology for my band's stage shows, but as I don't own
a MIDI‑controlled video presenter or projector, I can only assume that
this works as it should.

Finally, the Jupiter 80 incorporates a USB‑based song
player and recorder. Copy a suitable file (or files) to a USB drive and
stick it in the USB slot on the control panel, press the Song button and
you're ready to go.

In playback mode, you can fast‑forward and rewind,
loop within a file, chain files, alter the sound using the dedicated
four‑band EQ, and perform karaoke‑style centre cancellation. You can
also alter the playback speed and pitch, and although the algorithms for
these functions are not state‑of‑the‑art, they are adequate.

You can
even record audio back onto the memory stick as 44.1kHz, 16‑bit WAV
files, mixing your own performance with any audio being presented to the
USB or analogue audio inputs. I like this player; it's simple and
intuitive, and the independent speed and pitch controls will allow you
to work out all those fiddly Emerson Lake & Palmer piano solos that
have been bothering you for years [he's not kidding — Ed]. Nonetheless,
I'm unimpressed by the repeated exhortation that one should, "never
insert or remove a USB memory stick when the power is on”. Given that
the same port is used for saving and backing up the Jupiter 80's
memories, you would think that an 'eject' command and hot‑swapping would
be taken for granted, not precluded with dire warnings of the sky
falling in!

The Jupiter 80's keyboard is very pleasant to play; a good compromise
given the range of keyboard duties — from grand pianos, to organs, to
orchestral imitations, to synth solos — that it will be asked to
perform.
I suspect that this is in part because the size and weight of
the Jupiter 80 lends the keyboard a reassuring solidity, while the
instrument as a whole still remains more manageable than most synths
based on 88‑note piano‑weighted keybeds. This has to be a good thing for
an instrument designed to spend much of its life on the road. What's
more, the fact that the Jupiter 80 boots up markedly faster than
a V‑Synth or a Fantom makes it more suitable for live use in at least
one other sense: there must be nothing worse than standing on stage and
telling the audience you'll start playing again in a few minutes once
the keyboards have rebooted! Other things also suggest that Roland have
thought carefully about on‑stage use, and the area in which this is most
obvious is, perhaps, that of patch selection. The 27 large, colourful
buttons running behind the keyboard allow you to punch your choices from
54 predetermined Tones and Live Sets into the Parts of the
current Registration, and the Tone Remain function holds any existing
sound(s) until you release those keys, so there are no glitches on
changeover. What's more, you can specify the Tones and Live Sets that
are attached to each button, and you're not constrained to sounds that
conform to their names, so the system can be very flexible.

So now we come to the sound of the Jupiter 80. Starting
with the Supernatural Acoustic sounds, I have strong suspicions that,
rather than being a pure 'physical modelling' synth in the way that
I would historically have used that term, Supernatural Acoustic is
similar to Roland's Structured Adaptive Synthesis (SAS), which built its
piano and electric piano sounds using parametric models derived from
sample analysis. Given that a similar technology was recently used for
the V‑Piano, I emailed Roland to ask whether I was correct and whether
the piano sounds in the Jupiter 80 were the same as those found on other
Supernatural pianos such as the RD700 and FP7F. I hit a corporate brick
wall. The official response was, "we are unwilling to share this
information”, and when I asked for a list of the behaviours that
comprise Articulative Phrase Synthesis and Behavior Modeling Technology,
I obtained the same answer. Nonetheless, the Jupiter 80 is, as Roland
claim, a remarkably playable and expressive synthesizer, and many of its
Supernatural Acoustic sounds — such as the superb Clavinets, the
accordions and the excellent acoustic bass — are impressive, and the
subtle but sometimes important performance benefits of BMT and APS
shouldn't be overlooked.

Let's take an example. If you select one of the
acoustic guitars — say, 0035:SteelStr Guitar — and isolate this from
all the other sounds, you'll find that you can play it conventionally
and it will sound equivalent to patches from elsewhere. You could also
switch on the 'Strum Mode', so that chords are strummed in a realistic
fashion. But, again, this is nothing new. However, BMT becomes apparent
when you play two notes, either one or two semitones apart, quite hard
and almost simultaneously. You then hear a realistic glissando from the
first note to the second. If the interval is any greater, you obtain
a strum or a picked chord because BMT assumes than any interval above
two semitones is fretted separately or played on a different string.
Other easily audible examples of APS/BMT can be heard when you trill
rapidly and slur brass sounds, but there are some Tones for which the
effects are either so subtle that I'm missing them, or nothing is
implemented. Unfortunately, given Roland's reticence to discuss the
matter, I can't be any more informative.

Turning now to Supernatural Synth, I can only restate
how powerful the Jupiter 80 is in this department. You can stack three
Partials in a Tone and up to 10 Tones in a Registration to create some
monstrous patches combining analogue waveforms, Supersaw and PCMs! Or
course, it's much more sensible to create useful Tones and then layer
them in Live Sets to create some luscious sounds, but I suspect that the
main question on everyone's lips is, 'does it sound like a Jupiter 8?'.
As you might expect in a synth called the Jupiter 80, there are
numerous Tones, Live Sets and Registrations called 'Jupiter 8 something
or other', and a cynic might expect these to be nothing more than an
appeal to the gullible. However, notwithstanding a touch of aliasing at
the highest pitches, I found the Jupiter 80 to be capable of some
remarkably good imitations of the old lady. What's more, these
comparisons weren't against dim recollections of how a JP8 sounded when I heard one in a shop in Middlesbrough on a soggy afternoon in 1982... I
placed my Jupiter 8 next to the Jupiter 80 and compared them directly.
Of course there were differences, but in a blind test of some brass and
string patches, I couldn't tell which synth was generating which. This
was not what I had expected! Nonetheless, there is in my view one
significant shortcoming in Supernatural Synth; while you can affect the
loudness and brightness of a sound using aftertouch, you can't introduce
vibrato, tremolo or growl. For a synth that prides itself on its
performance capabilities, this seems a lamentable oversight.

You may have expected that something bearing the Jupiter name would
offer fistfuls of knobs and sliders, and a signal path harkening back to
the heyday of analogue synthesis. The Jupiter 80 does neither. So
should we conclude that its name and colour scheme are no more than
a cynical marketing exercise designed to drag cash out of the wallets of
the unwary? Certainly, it bears no more relation to a Jupiter 8 than
a Juno Stage bears to a Juno 60, so it's easy to leap to this
conclusion. But you must also remember that the Jupiter name only
assumed its current cachet some time after the original series had been
superseded by the JX8P and the Super JX10. In 1981, the Jupiter 8 was
merely Roland's interpretation of the current state of the art, designed
to compete head‑to‑head with the Prophet 5 and Oberheim OBX, so in the
sense that the Jupiter 80 is a performance synth based on the latest
technology, its name is not inappropriate. Nonetheless, it's going to
continue to annoy a lot of people.

Such issues aside, I'm relieved that Roland let me have
the pre‑release Jupiter 80 for such an extended period, because at the
start of this review I couldn't understand why they had designed such
a strange synth. But, as I learned how to approach its Tones, Live Sets
and Registrations, and as I began to work with what its effects
structure could do rather than complain about what it could not,
I started to discover what a remarkably expressive musical instrument
the Jupiter 80 is. I also began to realise that, had it been
manufactured elsewhere, somebody in the marketing department would have
been bouncing up and down and proclaiming loudly the multiple physical
models that comprise Supernatural Acoustic, whereas Roland have been
commendably conservative in their lack of hyperbole.

Of course, the Jupiter 80 is not for everybody, and if
you need a workstation capable of providing a dozen splits with
multitimbral effects assignments, you're looking at the wrong
instrument. But if you're after something that provides some top‑notch
piano and orchestral sounds, a remarkably powerful VA synth that can
imitate the best of the real thing, and the ability to build these into
complex, involving sonic structures, the Jupiter 80 has a lot going for
it. Sitting somewhere between a preset stage piano/organ/synth and
a fully featured workstation, it's a brave design, and — like me —
potential purchasers need to take the time to overcome their
preconceptions of what they think it should be, and begin to appreciate
it for what it really is. .